search for: _sip

Hi, We are using Asterisk 1.6.2 and it is continually failing to resolve Verizon SRV and sending following message, WARNING[23042]: acl.c:392 ast_get_ip_or_srv: Unable to lookup 'whsvoip.globalipcom.com' DNS settings on OS level is working fine. Can anyone have an idea about it? Regards, Faisal Hanif

Hello list. When using SIP DNS SRV to define a production Asterisk server with high priority and a backup Asterisk server with a lower priority on this DNS-server, will this work as follow : - production server is reachable, so registration of the IP-phone goes to this server - production server is unreachable, so registration goes to the backup Asterisk server - production server is

...for target 'iptel.org' res_pjsip/pjsip_resolver.c:177 sip_resolve_add: [0x7f4e740564e8] Added target 'iptel.org' with record type '35', transport 'Unspecified', and port '0' res_pjsip/pjsip_resolver.c:177 sip_resolve_add: [0x7f4e740564e8] Added target '_sips._tcp.iptel.org' with record type '33', transport 'TLS', and port '5061' res_pjsip/pjsip_resolver.c:177 sip_resolve_add: [0x7f4e740564e8] Added target '_sip._tcp.iptel.org' with record type '33', transport 'TCP', and port '5060' res_pjs...

...Engine : asterisk Codec Order : (ulaw:20,gsm:20) Auto-Framing: No tleilax*CLI> tleilax*CLI> exit tleilax:~ # tleilax:~ # exit logout Connection to tleilax closed. thufir at doge:~$ thufir at doge:~$ sudo sipsak -vv -s sip:123 at tleilax [sudo] password for thufir: No SRV record: _sip._tcp.tleilax No SRV record: _sip._udp.tleilax using A record: tleilax message received: SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.1.1:55238;branch=z9hG4bK.3e59b63f;alias;received=192.168.1.3;rport=55238 From: sip:sipsak at 127.0.1.1:55238;tag=1e6fe4eb To: sip:123 at tleilax;tag=as7dc4727d Call-ID: 51...

...sk Version: 1.4.2 Asterisk GUI Version: 2.0 The system was completely set up using the Asterisk GUI with a couple tweaks in users.conf that via:talk wants. Here is what happens: 1. Asterisk verifies connection to the server and we get this. (CLI output) -- ast_get_srv: SRV lookup for '_sip._udp.galvatron.vtnoc.net' mapped to host galvatron.vtnoc.net, port 5060 -- ast_get_srv: SRV lookup for '_sip._udp.galvatron.vtnoc.net' mapped to host optimusprime.vtnoc.net, port 5060 -- ast_get_srv: SRV lookup for '_sip._udp.galvatron.vtnoc.net' mapped to host megatr...

...risk.company.com. IN A 192.168.0.1 asterisk.company.com. IN A 192.168.0.2 asterisk.company.com. IN A 192.168.0.3 2. and normal A records for the servers like this: a1.company.com. IN A 192.168.0.1 a2.company.com. IN A 192.168.0.2 a3.company.com. IN A 192.168.0.3 3. and srv records like so: _sip._udp.company.com IN SRV 20 0 5060 a1.company.com _sip._udp.company.com IN SRV 30 0 5060 a2.company.com _sip._udp.company.com IN SRV 40 0 5060 a2.company.com 4. and configure all my sip clinets to register to asterisk.company.com 5. and tell the 3rd-party sip<-->pstn gateway to use srv rec...

Hi, SRV records have been working fine until my hoster decided to upgrade BIND. working wrong syntax: _sip._udp SRV 10 10 5050 mydyndns. correct syntax: _sip._udp IN SRV 10 10 5050 mydyndns. That kicked of one of my domains completly caused by a syntax error that did no harm to the previous version After inserting the missing "IN" the zonefile loaded but now I can'...

...00 secs Sess-Min-SE : 90 secs RTP Engine : asterisk Codec Order : (ulaw:20,gsm:20) Auto-Framing: No tleilax*CLI> which would make the URI sip:thufir101 at tleilax.bounceme.net ? thufir at doge:~$ thufir at doge:~$ sudo sipsak -vv -s sip:thufir101 at tleilax No SRV record: _sip._tcp.tleilax No SRV record: _sip._udp.tleilax using A record: tleilax message received: SIP/2.0 200 OK CSeq: 1 OPTIONS Via: SIP/2.0/UDP 127.0.1.1:52173;branch=z9hG4bK.4ca3965f;rport=52173;alias;received=192.168.1.3 User-Agent: Ekiga/4.0.1 From: sip:sipsak at 127.0.1.1:52173;tag=631bb564 Call-ID:...

...ell I do not ask those who only guess, but those who know what is > asterisk expected to do when internet connectivity goes down. I did not had > a chance to make internet not to work yet, since it is needed. But > inspecting dns logs I found out that there started to be resolving for > _sip._tcp and _sip._udp records for the provider's server. So apparently > making hosts record make asterisk happy when everything works, but when > there is a communication problem then it falls back to searching for srv > records. At least it seems to be so for now. Moreover I found out t...

...gt; Auto-Framing: No > > tleilax*CLI> > tleilax*CLI> exit > tleilax:~ # > tleilax:~ # exit > logout > Connection to tleilax closed. > thufir at doge:~$ > thufir at doge:~$ sudo sipsak -vv -s sip:123 at tleilax > [sudo] password for thufir: > No SRV record: _sip._tcp.tleilax > No SRV record: _sip._udp.tleilax > using A record: tleilax > > message received: > SIP/2.0 200 OK > Via: SIP/2.0/UDP > 127.0.1.1:55238;branch=z9hG4bK.3e59b63f;alias;received=192.168.1.3;rport=55238 > From: sip:sipsak at 127.0.1.1:55238;tag=1e6fe4eb > To: si...

Hello, well I do not ask those who only guess, but those who know what is asterisk expected to do when internet connectivity goes down. I did not had a chance to make internet not to work yet, since it is needed. But inspecting dns logs I found out that there started to be resolving for _sip._tcp and _sip._udp records for the provider's server. So apparently making hosts record make asterisk happy when everything works, but when there is a communication problem then it falls back to searching for srv records. At least it seems to be so for now. Moreover I found out this old thread:...

...9;ve tried it from an ekiga.net account and an sip2sip.info account. What could be wrong? I would expect incoming traffic on port 5060 UDP... The account is "paul at vandervlis.nl". This should connect trought DNS to the machine xen8.vandervlis.nl: ------- paul at server2:~$ host -t SRV _sip._udp.vandervlis.nl _sip._udp.vandervlis.nl has SRV record 0 5 5060 xen8.vandervlis.nl. ------- Is here maybe somebody with an idea, or a way to debug this? Maybe with a nice Linux commandline tool? With regards, Paul van der Vlis. -- http://www.vandervlis.nl/

...> 404. It probably has to do something with your context/dialplan. on tleilax: tleilax*CLI> tleilax*CLI> sip set debug on SIP Debugging enabled tleilax*CLI> on doge: thufir at doge:~$ thufir at doge:~$ sudo sipsak -vv -s sip:devries at tleilax -m "hi" No SRV record: _sip._tcp.tleilax No SRV record: _sip._udp.tleilax using A record: tleilax Max-Forwards set to 0 message received: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 127.0.1.1:56377;branch=z9hG4bK.0edaada3;alias;received=192.168.1.3;rport=56377 From: sip:sipsak at 127.0.1.1:56377;tag=6b540010 To: sip:devries at t...

Given the following SRV records: _sip._udp.tollfree.sip-happens.com. 38400 IN SRV 10 0 5060 sometimes.sip-happens.com. _sip._udp.tollfree.sip-happens.com. 38400 IN SRV 20 0 5070 ares.sip-happens.com. Why is asterisk (1.4.17) not honouring the priority and not failing over to using other records when a connection fails? For a given ca...

...or which services. > > We have such records in our DNS zone. They look like this: > > ;# Configure sip/sips service records (VOIP) > ;HOST TTL CLASS TYPE ORDER PREF FLAGS SERVICE REGEXP REPLACEMENT > > 300 IN NAPTR 50 50 "s" "SIPS+D2T" "" _sips._tcp.harte-lyne.ca. > > 300 IN NAPTR 90 50 "s" "SIP+D2T" "" _sip._tcp.harte-lyne.ca. > > 300 IN NAPTR 100 50 "s" "SIP+D2U" "" _sip._udp.harte-lyne.ca. > > ;HOST TTL CLASS TYPE ORDER PREF PORT TARGET > &g...

All, If I have video phones behind an asterisk server (with 2 network cards) and all the phones have extensions. Internally everything works great. Now for people that want to call my video phones external to my office is there a way to do that? On the extenal persons phone enter an IP/EXTEN where IP is my server and not the phone? Can that work? Would I have to have PUBLIC IP address for every

...ports the Snom will listen on for which services. We have such records in our DNS zone. They look like this: ;# Configure sip/sips service records (VOIP) ;HOST TTL CLASS TYPE ORDER PREF FLAGS SERVICE REGEXP REPLACEMENT 300 IN NAPTR 50 50 "s" "SIPS+D2T" "" _sips._tcp.harte-lyne.ca. 300 IN NAPTR 90 50 "s" "SIP+D2T" "" _sip._tcp.harte-lyne.ca. 300 IN NAPTR 100 50 "s" "SIP+D2U" "" _sip._udp.harte-lyne.ca. ;HOST TTL CLASS TYPE ORDER PREF PORT TARGET _sips._tcp.harte-lyne.ca. 300 IN...

Hey gang, I'm trying to setup the ability to dial a SIP user via their email address. I'm using SJPhone as my tester UA, but most clients will be using XTen Pro. I added an SRV DNS entry into our zone, and it returns: ; <<>> DiG 9.2.1 <<>> SRV _sip._udp.cytelcom.com ;; global options: printcmd ;; Got answer: ;; ->>HEADER<<- opcode: QUERY, status: NOERROR, id: 47390 ;; flags: qr rd ra; QUERY: 1, ANSWER: 1, AUTHORITY: 2, ADDITIONAL: 3 ;; QUESTION SECTION: ;_sip._udp.cytelcom.com. IN SRV ;; ANSWER SECTION: _sip...

...n Asterisk PBX recently and I encountered the following problem: When [mac address]-registration.cfg file includes the FQDN of the Asterisk PBX for the Polycom SoundPoint 501 phones it will not (even try to) register with the Asterisk PBX unless the DNS (it asks) successfully resolves the name: _sip._udp.[Asterisk FQDN]. Did this happen to anyone else? PS - The application version running on the SoundPoint 501s is 1.6.7.0098 TIA Paolo

...m' [Aug 27 07:51:36] DEBUG[595] res_pjsip/pjsip_resolver.c: [0x7f75282fe7f8] Added target 'rpi6.in.xorcom.com' with record type '35', transport 'Unspecified', and port '0' [Aug 27 07:51:36] DEBUG[595] res_pjsip/pjsip_resolver.c: [0x7f75282fe7f8] Added target '_sips._tcp.rpi6.in.xorcom.com' with record type '33', transport 'TLS', and port '5061' [Aug 27 07:51:36] DEBUG[595] res_pjsip/pjsip_resolver.c: [0x7f75282fe7f8] Added target '_sip._tcp.rpi6.in.xorcom.com' with record type '33', transport 'TCP', and por...